I just installed a TE820 octal span T1 card, and its not showing up in dahdi_hardware output.This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine.Is there some kernel module that..
I experience random crash of machine (full hang, requiring a hard reset)after trying to test run Asterisk 11.The machine is a CentOS 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installedAnyone experie..
I have noticed some occasional one way audio on a specific sub set of calls in my system. First, let me be more specific about what I mean about occasional one way audio. Unlike most of the posts Ive seen (where the end fix was either NATing or RTP issu..
we would really like to be able to invite a third and fourth partyto our current one-on-one call. At the moment, we have to agree to dial into MeetMe 10 minutes later, then make calls to the third parties, and hope it all works out.I have found a cou..
Lionel BEAUDOIN wrote:Hola,Please ensure you have followed the instructions at https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support to set up the Asterisk side of things for WebSocket.You dont need to use your own copy of sipml5. Po..
I am using 1.4.43 currently.I am using the AMI to originate a call over a SIP Trunk to my cell XXX506YYYY. works fine. when the call is active I do a core show channels concise and I get:SIP/testsystem-00000ad0!smvoice-dialout!callprogress!4!Up!AGI!smvoice!0!!3!24!(None..
we are finally going to redesign our Asterisk-Setup, which has grown quite complex. We have five sites with a total of 400 users, 15 SIPregistrations and 3 IAX registrations. We do not use any VoIP-hardware, so its all software-based. But we make he..